// Copyright (c) 2012 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.

#include "media/audio/win/audio_low_latency_input_win.h"

#include <memory>

#include "base/logging.h"
#include "base/strings/utf_string_conversions.h"
#include "base/trace_event/trace_event.h"
#include "media/audio/audio_device_description.h"
#include "media/audio/win/audio_manager_win.h"
#include "media/audio/win/avrt_wrapper_win.h"
#include "media/audio/win/core_audio_util_win.h"
#include "media/base/audio_bus.h"

using base::win::ScopedCOMInitializer;
using base::win::ScopedComPtr;

namespace media {

WASAPIAudioInputStream::WASAPIAudioInputStream(AudioManagerWin* manager,
    const AudioParameters& params,
    const std::string& device_id)
    : manager_(manager)
    , capture_thread_(NULL)
    , opened_(false)
    , started_(false)
    , frame_size_(0)
    , packet_size_frames_(0)
    , packet_size_bytes_(0)
    , endpoint_buffer_size_frames_(0)
    , device_id_(device_id)
    , perf_count_to_100ns_units_(0.0)
    , ms_to_frame_count_(0.0)
    , sink_(NULL)
    , audio_bus_(media::AudioBus::Create(params))
    , mute_done_(false)
{
    DCHECK(manager_);

    // Load the Avrt DLL if not already loaded. Required to support MMCSS.
    bool avrt_init = avrt::Initialize();
    DCHECK(avrt_init) << "Failed to load the Avrt.dll";

    // Set up the desired capture format specified by the client.
    format_.nSamplesPerSec = params.sample_rate();
    format_.wFormatTag = WAVE_FORMAT_PCM;
    format_.wBitsPerSample = params.bits_per_sample();
    format_.nChannels = params.channels();
    format_.nBlockAlign = (format_.wBitsPerSample / 8) * format_.nChannels;
    format_.nAvgBytesPerSec = format_.nSamplesPerSec * format_.nBlockAlign;
    format_.cbSize = 0;

    // Size in bytes of each audio frame.
    frame_size_ = format_.nBlockAlign;
    // Store size of audio packets which we expect to get from the audio
    // endpoint device in each capture event.
    packet_size_frames_ = params.GetBytesPerBuffer() / format_.nBlockAlign;
    packet_size_bytes_ = params.GetBytesPerBuffer();
    DVLOG(1) << "Number of bytes per audio frame  : " << frame_size_;
    DVLOG(1) << "Number of audio frames per packet: " << packet_size_frames_;

    // All events are auto-reset events and non-signaled initially.

    // Create the event which the audio engine will signal each time
    // a buffer becomes ready to be processed by the client.
    audio_samples_ready_event_.Set(CreateEvent(NULL, FALSE, FALSE, NULL));
    DCHECK(audio_samples_ready_event_.IsValid());

    // Create the event which will be set in Stop() when capturing shall stop.
    stop_capture_event_.Set(CreateEvent(NULL, FALSE, FALSE, NULL));
    DCHECK(stop_capture_event_.IsValid());

    ms_to_frame_count_ = static_cast<double>(params.sample_rate()) / 1000.0;

    LARGE_INTEGER performance_frequency;
    if (QueryPerformanceFrequency(&performance_frequency)) {
        perf_count_to_100ns_units_ = (10000000.0 / static_cast<double>(performance_frequency.QuadPart));
    } else {
        DLOG(ERROR) << "High-resolution performance counters are not supported.";
    }
}

WASAPIAudioInputStream::~WASAPIAudioInputStream()
{
    DCHECK(CalledOnValidThread());
}

bool WASAPIAudioInputStream::Open()
{
    DCHECK(CalledOnValidThread());
    // Verify that we are not already opened.
    if (opened_)
        return false;

    // Obtain a reference to the IMMDevice interface of the capturing
    // device with the specified unique identifier or role which was
    // set at construction.
    HRESULT hr = SetCaptureDevice();
    if (FAILED(hr))
        return false;

    // Obtain an IAudioClient interface which enables us to create and initialize
    // an audio stream between an audio application and the audio engine.
    hr = ActivateCaptureDevice();
    if (FAILED(hr))
        return false;

        // Retrieve the stream format which the audio engine uses for its internal
        // processing/mixing of shared-mode streams. This function call is for
        // diagnostic purposes only and only in debug mode.
#ifndef NDEBUG
    hr = GetAudioEngineStreamFormat();
#endif

    // Verify that the selected audio endpoint supports the specified format
    // set during construction.
    if (!DesiredFormatIsSupported())
        return false;

    // Initialize the audio stream between the client and the device using
    // shared mode and a lowest possible glitch-free latency.
    hr = InitializeAudioEngine();

    opened_ = SUCCEEDED(hr);
    return opened_;
}

void WASAPIAudioInputStream::Start(AudioInputCallback* callback)
{
    DCHECK(CalledOnValidThread());
    DCHECK(callback);
    DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully";
    if (!opened_)
        return;

    if (started_)
        return;

    if (device_id_ == AudioDeviceDescription::kLoopbackWithMuteDeviceId && system_audio_volume_) {
        BOOL muted = false;
        system_audio_volume_->GetMute(&muted);

        // If the system audio is muted at the time of capturing, then no need to
        // mute it again, and later we do not unmute system audio when stopping
        // capturing.
        if (!muted) {
            system_audio_volume_->SetMute(true, NULL);
            mute_done_ = true;
        }
    }

    DCHECK(!sink_);
    sink_ = callback;

    // Starts periodic AGC microphone measurements if the AGC has been enabled
    // using SetAutomaticGainControl().
    StartAgc();

    // Create and start the thread that will drive the capturing by waiting for
    // capture events.
    capture_thread_ = new base::DelegateSimpleThread(
        this, "wasapi_capture_thread",
        base::SimpleThread::Options(base::ThreadPriority::REALTIME_AUDIO));
    capture_thread_->Start();

    // Start streaming data between the endpoint buffer and the audio engine.
    HRESULT hr = audio_client_->Start();
    DLOG_IF(ERROR, FAILED(hr)) << "Failed to start input streaming.";

    if (SUCCEEDED(hr) && audio_render_client_for_loopback_.get())
        hr = audio_render_client_for_loopback_->Start();

    started_ = SUCCEEDED(hr);
}

void WASAPIAudioInputStream::Stop()
{
    DCHECK(CalledOnValidThread());
    DVLOG(1) << "WASAPIAudioInputStream::Stop()";
    if (!started_)
        return;

    // We have muted system audio for capturing, so we need to unmute it when
    // capturing stops.
    if (device_id_ == AudioDeviceDescription::kLoopbackWithMuteDeviceId && mute_done_) {
        DCHECK(system_audio_volume_);
        if (system_audio_volume_) {
            system_audio_volume_->SetMute(false, NULL);
            mute_done_ = false;
        }
    }

    // Stops periodic AGC microphone measurements.
    StopAgc();

    // Shut down the capture thread.
    if (stop_capture_event_.IsValid()) {
        SetEvent(stop_capture_event_.Get());
    }

    // Stop the input audio streaming.
    HRESULT hr = audio_client_->Stop();
    if (FAILED(hr)) {
        LOG(ERROR) << "Failed to stop input streaming.";
    }

    // Wait until the thread completes and perform cleanup.
    if (capture_thread_) {
        SetEvent(stop_capture_event_.Get());
        capture_thread_->Join();
        capture_thread_ = NULL;
    }

    started_ = false;
    sink_ = NULL;
}

void WASAPIAudioInputStream::Close()
{
    DVLOG(1) << "WASAPIAudioInputStream::Close()";
    // It is valid to call Close() before calling open or Start().
    // It is also valid to call Close() after Start() has been called.
    Stop();

    // Inform the audio manager that we have been closed. This will cause our
    // destruction.
    manager_->ReleaseInputStream(this);
}

double WASAPIAudioInputStream::GetMaxVolume()
{
    // Verify that Open() has been called succesfully, to ensure that an audio
    // session exists and that an ISimpleAudioVolume interface has been created.
    DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully";
    if (!opened_)
        return 0.0;

    // The effective volume value is always in the range 0.0 to 1.0, hence
    // we can return a fixed value (=1.0) here.
    return 1.0;
}

void WASAPIAudioInputStream::SetVolume(double volume)
{
    DVLOG(1) << "SetVolume(volume=" << volume << ")";
    DCHECK(CalledOnValidThread());
    DCHECK_GE(volume, 0.0);
    DCHECK_LE(volume, 1.0);

    DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully";
    if (!opened_)
        return;

    // Set a new master volume level. Valid volume levels are in the range
    // 0.0 to 1.0. Ignore volume-change events.
    HRESULT hr = simple_audio_volume_->SetMasterVolume(static_cast<float>(volume), NULL);
    if (FAILED(hr))
        DLOG(WARNING) << "Failed to set new input master volume.";

    // Update the AGC volume level based on the last setting above. Note that,
    // the volume-level resolution is not infinite and it is therefore not
    // possible to assume that the volume provided as input parameter can be
    // used directly. Instead, a new query to the audio hardware is required.
    // This method does nothing if AGC is disabled.
    UpdateAgcVolume();
}

double WASAPIAudioInputStream::GetVolume()
{
    DCHECK(opened_) << "Open() has not been called successfully";
    if (!opened_)
        return 0.0;

    // Retrieve the current volume level. The value is in the range 0.0 to 1.0.
    float level = 0.0f;
    HRESULT hr = simple_audio_volume_->GetMasterVolume(&level);
    if (FAILED(hr))
        DLOG(WARNING) << "Failed to get input master volume.";

    return static_cast<double>(level);
}

bool WASAPIAudioInputStream::IsMuted()
{
    DCHECK(opened_) << "Open() has not been called successfully";
    DCHECK(CalledOnValidThread());
    if (!opened_)
        return false;

    // Retrieves the current muting state for the audio session.
    BOOL is_muted = FALSE;
    HRESULT hr = simple_audio_volume_->GetMute(&is_muted);
    if (FAILED(hr))
        DLOG(WARNING) << "Failed to get input master volume.";

    return is_muted != FALSE;
}

void WASAPIAudioInputStream::Run()
{
    ScopedCOMInitializer com_init(ScopedCOMInitializer::kMTA);

    // Enable MMCSS to ensure that this thread receives prioritized access to
    // CPU resources.
    DWORD task_index = 0;
    HANDLE mm_task = avrt::AvSetMmThreadCharacteristics(L"Pro Audio",
        &task_index);
    bool mmcss_is_ok = (mm_task && avrt::AvSetMmThreadPriority(mm_task, AVRT_PRIORITY_CRITICAL));
    if (!mmcss_is_ok) {
        // Failed to enable MMCSS on this thread. It is not fatal but can lead
        // to reduced QoS at high load.
        DWORD err = GetLastError();
        LOG(WARNING) << "Failed to enable MMCSS (error code=" << err << ").";
    }

    // Allocate a buffer with a size that enables us to take care of cases like:
    // 1) The recorded buffer size is smaller, or does not match exactly with,
    //    the selected packet size used in each callback.
    // 2) The selected buffer size is larger than the recorded buffer size in
    //    each event.
    size_t buffer_frame_index = 0;
    size_t capture_buffer_size = std::max(
        2 * endpoint_buffer_size_frames_ * frame_size_,
        2 * packet_size_frames_ * frame_size_);
    std::unique_ptr<uint8_t[]> capture_buffer(new uint8_t[capture_buffer_size]);

    LARGE_INTEGER now_count = {};
    bool recording = true;
    bool error = false;
    double volume = GetVolume();
    HANDLE wait_array[2] = { stop_capture_event_.Get(), audio_samples_ready_event_.Get() };

    base::win::ScopedComPtr<IAudioClock> audio_clock;
    audio_client_->GetService(__uuidof(IAudioClock), audio_clock.ReceiveVoid());

    while (recording && !error) {
        HRESULT hr = S_FALSE;

        // Wait for a close-down event or a new capture event.
        DWORD wait_result = WaitForMultipleObjects(2, wait_array, FALSE, INFINITE);
        switch (wait_result) {
        case WAIT_FAILED:
            error = true;
            break;
        case WAIT_OBJECT_0 + 0:
            // |stop_capture_event_| has been set.
            recording = false;
            break;
        case WAIT_OBJECT_0 + 1: {
            TRACE_EVENT0("audio", "WASAPIAudioInputStream::Run_0");
            // |audio_samples_ready_event_| has been set.
            BYTE* data_ptr = NULL;
            UINT32 num_frames_to_read = 0;
            DWORD flags = 0;
            UINT64 device_position = 0;
            UINT64 first_audio_frame_timestamp = 0;

            // Retrieve the amount of data in the capture endpoint buffer,
            // replace it with silence if required, create callbacks for each
            // packet and store non-delivered data for the next event.
            hr = audio_capture_client_->GetBuffer(&data_ptr,
                &num_frames_to_read,
                &flags,
                &device_position,
                &first_audio_frame_timestamp);
            if (FAILED(hr)) {
                DLOG(ERROR) << "Failed to get data from the capture buffer";
                continue;
            }

            if (audio_clock) {
                // The reported timestamp from GetBuffer is not as reliable as the
                // clock from the client.  We've seen timestamps reported for
                // USB audio devices, be off by several days.  Furthermore we've
                // seen them jump back in time every 2 seconds or so.
                audio_clock->GetPosition(
                    &device_position, &first_audio_frame_timestamp);
            }

            if (num_frames_to_read != 0) {
                size_t pos = buffer_frame_index * frame_size_;
                size_t num_bytes = num_frames_to_read * frame_size_;
                DCHECK_GE(capture_buffer_size, pos + num_bytes);

                if (flags & AUDCLNT_BUFFERFLAGS_SILENT) {
                    // Clear out the local buffer since silence is reported.
                    memset(&capture_buffer[pos], 0, num_bytes);
                } else {
                    // Copy captured data from audio engine buffer to local buffer.
                    memcpy(&capture_buffer[pos], data_ptr, num_bytes);
                }

                buffer_frame_index += num_frames_to_read;
            }

            hr = audio_capture_client_->ReleaseBuffer(num_frames_to_read);
            DLOG_IF(ERROR, FAILED(hr)) << "Failed to release capture buffer";

            // Derive a delay estimate for the captured audio packet.
            // The value contains two parts (A+B), where A is the delay of the
            // first audio frame in the packet and B is the extra delay
            // contained in any stored data. Unit is in audio frames.
            QueryPerformanceCounter(&now_count);
            // first_audio_frame_timestamp will be 0 if we didn't get a timestamp.
            double audio_delay_frames = first_audio_frame_timestamp == 0 ? num_frames_to_read : ((perf_count_to_100ns_units_ * now_count.QuadPart - first_audio_frame_timestamp) / 10000.0) * ms_to_frame_count_ + buffer_frame_index - num_frames_to_read;

            // Get a cached AGC volume level which is updated once every second
            // on the audio manager thread. Note that, |volume| is also updated
            // each time SetVolume() is called through IPC by the render-side AGC.
            GetAgcVolume(&volume);

            // Deliver captured data to the registered consumer using a packet
            // size which was specified at construction.
            uint32_t delay_frames = static_cast<uint32_t>(audio_delay_frames + 0.5);
            while (buffer_frame_index >= packet_size_frames_) {
                // Copy data to audio bus to match the OnData interface.
                uint8_t* audio_data = reinterpret_cast<uint8_t*>(capture_buffer.get());
                audio_bus_->FromInterleaved(
                    audio_data, audio_bus_->frames(), format_.wBitsPerSample / 8);

                // Deliver data packet, delay estimation and volume level to
                // the user.
                sink_->OnData(
                    this, audio_bus_.get(), delay_frames * frame_size_, volume);

                // Store parts of the recorded data which can't be delivered
                // using the current packet size. The stored section will be used
                // either in the next while-loop iteration or in the next
                // capture event.
                // TODO(tommi): If this data will be used in the next capture
                // event, we will report incorrect delay estimates because
                // we'll use the one for the captured data that time around
                // (i.e. in the future).
                memmove(&capture_buffer[0],
                    &capture_buffer[packet_size_bytes_],
                    (buffer_frame_index - packet_size_frames_) * frame_size_);

                DCHECK_GE(buffer_frame_index, packet_size_frames_);
                buffer_frame_index -= packet_size_frames_;
                if (delay_frames > packet_size_frames_) {
                    delay_frames -= packet_size_frames_;
                } else {
                    delay_frames = 0;
                }
            }
        } break;
        default:
            error = true;
            break;
        }
    }

    if (recording && error) {
        // TODO(henrika): perhaps it worth improving the cleanup here by e.g.
        // stopping the audio client, joining the thread etc.?
        NOTREACHED() << "WASAPI capturing failed with error code "
                     << GetLastError();
    }

    // Disable MMCSS.
    if (mm_task && !avrt::AvRevertMmThreadCharacteristics(mm_task)) {
        PLOG(WARNING) << "Failed to disable MMCSS";
    }
}

void WASAPIAudioInputStream::HandleError(HRESULT err)
{
    NOTREACHED() << "Error code: " << err;
    if (sink_)
        sink_->OnError(this);
}

HRESULT WASAPIAudioInputStream::SetCaptureDevice()
{
    DCHECK(!endpoint_device_.get());

    ScopedComPtr<IMMDeviceEnumerator> enumerator;
    HRESULT hr = enumerator.CreateInstance(__uuidof(MMDeviceEnumerator),
        NULL, CLSCTX_INPROC_SERVER);
    if (FAILED(hr))
        return hr;

    // Retrieve the IMMDevice by using the specified role or the specified
    // unique endpoint device-identification string.

    if (device_id_ == AudioDeviceDescription::kDefaultDeviceId) {
        // Retrieve the default capture audio endpoint for the specified role.
        // Note that, in Windows Vista, the MMDevice API supports device roles
        // but the system-supplied user interface programs do not.
        hr = enumerator->GetDefaultAudioEndpoint(eCapture, eConsole,
            endpoint_device_.Receive());
    } else if (device_id_ == AudioDeviceDescription::kCommunicationsDeviceId) {
        hr = enumerator->GetDefaultAudioEndpoint(eCapture, eCommunications,
            endpoint_device_.Receive());
    } else if (device_id_ == AudioDeviceDescription::kLoopbackWithMuteDeviceId) {
        // Capture the default playback stream.
        hr = enumerator->GetDefaultAudioEndpoint(eRender, eConsole,
            endpoint_device_.Receive());

        endpoint_device_->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
            system_audio_volume_.ReceiveVoid());
    } else if (device_id_ == AudioDeviceDescription::kLoopbackInputDeviceId) {
        // Capture the default playback stream.
        hr = enumerator->GetDefaultAudioEndpoint(eRender, eConsole,
            endpoint_device_.Receive());
    } else {
        hr = enumerator->GetDevice(base::UTF8ToUTF16(device_id_).c_str(),
            endpoint_device_.Receive());
    }

    if (FAILED(hr))
        return hr;

    // Verify that the audio endpoint device is active, i.e., the audio
    // adapter that connects to the endpoint device is present and enabled.
    DWORD state = DEVICE_STATE_DISABLED;
    hr = endpoint_device_->GetState(&state);
    if (FAILED(hr))
        return hr;

    if (!(state & DEVICE_STATE_ACTIVE)) {
        DLOG(ERROR) << "Selected capture device is not active.";
        hr = E_ACCESSDENIED;
    }

    return hr;
}

HRESULT WASAPIAudioInputStream::ActivateCaptureDevice()
{
    // Creates and activates an IAudioClient COM object given the selected
    // capture endpoint device.
    HRESULT hr = endpoint_device_->Activate(__uuidof(IAudioClient),
        CLSCTX_INPROC_SERVER,
        NULL,
        audio_client_.ReceiveVoid());
    return hr;
}

HRESULT WASAPIAudioInputStream::GetAudioEngineStreamFormat()
{
    HRESULT hr = S_OK;
#ifndef NDEBUG
    // The GetMixFormat() method retrieves the stream format that the
    // audio engine uses for its internal processing of shared-mode streams.
    // The method always uses a WAVEFORMATEXTENSIBLE structure, instead
    // of a stand-alone WAVEFORMATEX structure, to specify the format.
    // An WAVEFORMATEXTENSIBLE structure can specify both the mapping of
    // channels to speakers and the number of bits of precision in each sample.
    base::win::ScopedCoMem<WAVEFORMATEXTENSIBLE> format_ex;
    hr = audio_client_->GetMixFormat(
        reinterpret_cast<WAVEFORMATEX**>(&format_ex));

    // See http://msdn.microsoft.com/en-us/windows/hardware/gg463006#EFH
    // for details on the WAVE file format.
    WAVEFORMATEX format = format_ex->Format;
    DVLOG(2) << "WAVEFORMATEX:";
    DVLOG(2) << "  wFormatTags    : 0x" << std::hex << format.wFormatTag;
    DVLOG(2) << "  nChannels      : " << format.nChannels;
    DVLOG(2) << "  nSamplesPerSec : " << format.nSamplesPerSec;
    DVLOG(2) << "  nAvgBytesPerSec: " << format.nAvgBytesPerSec;
    DVLOG(2) << "  nBlockAlign    : " << format.nBlockAlign;
    DVLOG(2) << "  wBitsPerSample : " << format.wBitsPerSample;
    DVLOG(2) << "  cbSize         : " << format.cbSize;

    DVLOG(2) << "WAVEFORMATEXTENSIBLE:";
    DVLOG(2) << " wValidBitsPerSample: " << format_ex->Samples.wValidBitsPerSample;
    DVLOG(2) << " dwChannelMask      : 0x" << std::hex << format_ex->dwChannelMask;
    if (format_ex->SubFormat == KSDATAFORMAT_SUBTYPE_PCM)
        DVLOG(2) << " SubFormat          : KSDATAFORMAT_SUBTYPE_PCM";
    else if (format_ex->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)
        DVLOG(2) << " SubFormat          : KSDATAFORMAT_SUBTYPE_IEEE_FLOAT";
    else if (format_ex->SubFormat == KSDATAFORMAT_SUBTYPE_WAVEFORMATEX)
        DVLOG(2) << " SubFormat          : KSDATAFORMAT_SUBTYPE_WAVEFORMATEX";
#endif
    return hr;
}

bool WASAPIAudioInputStream::DesiredFormatIsSupported()
{
    // An application that uses WASAPI to manage shared-mode streams can rely
    // on the audio engine to perform only limited format conversions. The audio
    // engine can convert between a standard PCM sample size used by the
    // application and the floating-point samples that the engine uses for its
    // internal processing. However, the format for an application stream
    // typically must have the same number of channels and the same sample
    // rate as the stream format used by the device.
    // Many audio devices support both PCM and non-PCM stream formats. However,
    // the audio engine can mix only PCM streams.
    base::win::ScopedCoMem<WAVEFORMATEX> closest_match;
    HRESULT hr = audio_client_->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED,
        &format_,
        &closest_match);
    DLOG_IF(ERROR, hr == S_FALSE) << "Format is not supported "
                                  << "but a closest match exists.";
    return (hr == S_OK);
}

HRESULT WASAPIAudioInputStream::InitializeAudioEngine()
{
    DWORD flags;
    // Use event-driven mode only fo regular input devices. For loopback the
    // EVENTCALLBACK flag is specified when intializing
    // |audio_render_client_for_loopback_|.
    if (device_id_ == AudioDeviceDescription::kLoopbackInputDeviceId || device_id_ == AudioDeviceDescription::kLoopbackWithMuteDeviceId) {
        flags = AUDCLNT_STREAMFLAGS_LOOPBACK | AUDCLNT_STREAMFLAGS_NOPERSIST;
    } else {
        flags = AUDCLNT_STREAMFLAGS_EVENTCALLBACK | AUDCLNT_STREAMFLAGS_NOPERSIST;
    }

    // Initialize the audio stream between the client and the device.
    // We connect indirectly through the audio engine by using shared mode.
    // Note that, |hnsBufferDuration| is set of 0, which ensures that the
    // buffer is never smaller than the minimum buffer size needed to ensure
    // that glitches do not occur between the periodic processing passes.
    // This setting should lead to lowest possible latency.
    HRESULT hr = audio_client_->Initialize(
        AUDCLNT_SHAREMODE_SHARED, flags,
        0, // hnsBufferDuration
        0, &format_, device_id_ == AudioDeviceDescription::kCommunicationsDeviceId ? &kCommunicationsSessionId : nullptr);

    if (FAILED(hr))
        return hr;

    // Retrieve the length of the endpoint buffer shared between the client
    // and the audio engine. The buffer length determines the maximum amount
    // of capture data that the audio engine can read from the endpoint buffer
    // during a single processing pass.
    // A typical value is 960 audio frames <=> 20ms @ 48kHz sample rate.
    hr = audio_client_->GetBufferSize(&endpoint_buffer_size_frames_);
    if (FAILED(hr))
        return hr;

    DVLOG(1) << "endpoint buffer size: " << endpoint_buffer_size_frames_
             << " [frames]";

#ifndef NDEBUG
    // The period between processing passes by the audio engine is fixed for a
    // particular audio endpoint device and represents the smallest processing
    // quantum for the audio engine. This period plus the stream latency between
    // the buffer and endpoint device represents the minimum possible latency
    // that an audio application can achieve.
    // TODO(henrika): possibly remove this section when all parts are ready.
    REFERENCE_TIME device_period_shared_mode = 0;
    REFERENCE_TIME device_period_exclusive_mode = 0;
    HRESULT hr_dbg = audio_client_->GetDevicePeriod(
        &device_period_shared_mode, &device_period_exclusive_mode);
    if (SUCCEEDED(hr_dbg)) {
        DVLOG(1) << "device period: "
                 << static_cast<double>(device_period_shared_mode / 10000.0)
                 << " [ms]";
    }

    REFERENCE_TIME latency = 0;
    hr_dbg = audio_client_->GetStreamLatency(&latency);
    if (SUCCEEDED(hr_dbg)) {
        DVLOG(1) << "stream latency: " << static_cast<double>(latency / 10000.0)
                 << " [ms]";
    }
#endif

    // Set the event handle that the audio engine will signal each time a buffer
    // becomes ready to be processed by the client.
    //
    // In loopback case the capture device doesn't receive any events, so we
    // need to create a separate playback client to get notifications. According
    // to MSDN:
    //
    //   A pull-mode capture client does not receive any events when a stream is
    //   initialized with event-driven buffering and is loopback-enabled. To
    //   work around this, initialize a render stream in event-driven mode. Each
    //   time the client receives an event for the render stream, it must signal
    //   the capture client to run the capture thread that reads the next set of
    //   samples from the capture endpoint buffer.
    //
    // http://msdn.microsoft.com/en-us/library/windows/desktop/dd316551(v=vs.85).aspx
    if (device_id_ == AudioDeviceDescription::kLoopbackInputDeviceId || device_id_ == AudioDeviceDescription::kLoopbackWithMuteDeviceId) {
        hr = endpoint_device_->Activate(
            __uuidof(IAudioClient), CLSCTX_INPROC_SERVER, NULL,
            audio_render_client_for_loopback_.ReceiveVoid());
        if (FAILED(hr))
            return hr;

        hr = audio_render_client_for_loopback_->Initialize(
            AUDCLNT_SHAREMODE_SHARED,
            AUDCLNT_STREAMFLAGS_EVENTCALLBACK | AUDCLNT_STREAMFLAGS_NOPERSIST,
            0, 0, &format_, NULL);
        if (FAILED(hr))
            return hr;

        hr = audio_render_client_for_loopback_->SetEventHandle(
            audio_samples_ready_event_.Get());
    } else {
        hr = audio_client_->SetEventHandle(audio_samples_ready_event_.Get());
    }

    if (FAILED(hr))
        return hr;

    // Get access to the IAudioCaptureClient interface. This interface
    // enables us to read input data from the capture endpoint buffer.
    hr = audio_client_->GetService(__uuidof(IAudioCaptureClient),
        audio_capture_client_.ReceiveVoid());
    if (FAILED(hr))
        return hr;

    // Obtain a reference to the ISimpleAudioVolume interface which enables
    // us to control the master volume level of an audio session.
    hr = audio_client_->GetService(__uuidof(ISimpleAudioVolume),
        simple_audio_volume_.ReceiveVoid());
    return hr;
}

} // namespace media
